Pass time constant to bwe smoothing filter.

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 2cb12750211842bda5111e67dcba460fb6570800..e71304056b79c590fe7fa788850cc40d7e4586c6 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -98,6 +98,8 @@ struct ConfigHelper {
     // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec.codec_inst = kIsacCodec;
+    stream_config_.min_bitrate_bps = 10000;
+    stream_config_.max_bitrate_bps = 65000;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
@@ -391,5 +393,30 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
       helper.event_log());
 }
 
+TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(
+      helper.config(), helper.audio_state(), helper.worker_queue(),
+      helper.congestion_controller(), helper.bitrate_allocator(),
+      helper.event_log());
+  EXPECT_CALL(*helper.channel_proxy(),
+              SetBitrate(helper.config().max_bitrate_bps, _));
+  send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
+                               6000);
+}
+
+TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
+  ConfigHelper helper;
+  internal::AudioSendStream send_stream(
+      helper.config(), helper.audio_state(), helper.worker_queue(),
+      helper.congestion_controller(), helper.bitrate_allocator(),
+      helper.event_log());
+  EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
+  send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
+
+  EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 6000));
+  send_stream.OnBitrateUpdated(50000, 0.0, 50, 6000);
+}
+
 }  // namespace test
 }  // namespace webrtc